Video calls
Video calls work between the web phone, the Android app and desk video phones. They are off by default: video uses many times the bandwidth of audio.
Switching them on
- System → Asterisk settings → Video: Video calls on, then the video codec, the maximum resolution and frame rate. Apply the change.
- Each user needs the video phone mode (Users or My Phone → Call settings); it is on by default.
| Video codec | Use it when |
|---|---|
| VP8 (default) | Calls between browsers and the mobile apps. Every browser and every Android phone can do it. |
| H.264 | You also have desk video phones (Yealink, Grandstream GXV, Fanvil …), which mostly speak only H.264. Browsers handle it; the Android app needs a phone that can encode H.264, otherwise it talks without video. |
There is only one codec for the whole PBX on purpose. Asterisk passes video through without converting it, and a call whose two sides settled on different video codecs is hung up by Asterisk ("Don't know how to send format … packets"). With one codec everywhere a device that lacks it simply talks without video.
Using them
- Web phone: the camera button next to the dial field calls with video. An incoming video call shows Video next to Answer. During the call a window shows the other side with the own camera in the corner, with full screen and a camera on/off button in the top bar.
- Android app (1.0.65+): tap a colleague in Contacts or History and choose Video call; an incoming video call has a Video button next to Answer. During the call: camera on/off and front/back camera; the speaker is switched on.
- Answering with Answer (audio) still shows the caller's picture if they send one.
- Without a camera, with the camera busy or the permission refused, the call goes on with audio.
Notes
- Video calls are bridged by Asterisk; there is no video conference or recording of the picture.
- While video calls are on, the WebRTC endpoints have no RTP timeout: browsers send audio and video over one bundled port and Asterisk would see the video stream as silent and hang up after 30 s. Calls of a closed browser or app still end through SIP session timers.
- Video between two WebRTC users goes through the TURN relay like audio when they are behind NAT (see WebRTC, NAT and TURN); plan the bandwidth of the relay accordingly.